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/*
* ALSA input and output
* Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* ALSA input and output: input
* @author Luca Abeni ( lucabe72 email it )
* @author Benoit Fouet ( benoit fouet free fr )
* @author Nicolas George ( nicolas george normalesup org )
*
* This avdevice decoder can capture audio from an ALSA (Advanced
* Linux Sound Architecture) device.
*
* The filename parameter is the name of an ALSA PCM device capable of
* capture, for example "default" or "plughw:1"; see the ALSA documentation
* for naming conventions. The empty string is equivalent to "default".
*
* The capture period is set to the lower value available for the device,
* which gives a low latency suitable for real-time capture.
*
* The PTS are an Unix time in microsecond.
*
* Due to a bug in the ALSA library
* (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this
* decoder does not work with certain ALSA plugins, especially the dsnoop
* plugin.
*/
#include <alsa/asoundlib.h>
#include "libavutil/internal.h"
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "libavformat/internal.h"
#include "avdevice.h"
#include "alsa.h"
static av_cold int audio_read_header(AVFormatContext *s1)
{
AlsaData *s = s1->priv_data;
AVStream *st;
int ret;
enum AVCodecID codec_id;
st = avformat_new_stream(s1, NULL);
if (!st) {
av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
return AVERROR(ENOMEM);
}
codec_id = s1->audio_codec_id;
ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
&codec_id);
if (ret < 0) {
return AVERROR(EIO);
}
/* take real parameters */
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = codec_id;
st->codecpar->sample_rate = s->sample_rate;
st->codecpar->channels = s->channels;
st->codecpar->frame_size = s->frame_size;
avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
/* microseconds instead of seconds, MHz instead of Hz */
s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
s->period_size, 1.5E-6);
if (!s->timefilter)
goto fail;
return 0;
fail:
snd_pcm_close(s->h);
return AVERROR(EIO);
}
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
{
AlsaData *s = s1->priv_data;
int res;
int64_t dts;
snd_pcm_sframes_t delay = 0;
if (!s->pkt->data) {
int ret = av_new_packet(s->pkt, s->period_size * s->frame_size);
if (ret < 0)
return ret;
s->pkt->size = 0;
}
do {
while ((res = snd_pcm_readi(s->h, s->pkt->data + s->pkt->size, s->period_size - s->pkt->size / s->frame_size)) < 0) {
if (res == -EAGAIN) {
return AVERROR(EAGAIN);
}
s->pkt->size = 0;
if (ff_alsa_xrun_recover(s1, res) < 0) {
av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
snd_strerror(res));
return AVERROR(EIO);
}
ff_timefilter_reset(s->timefilter);
}
s->pkt->size += res * s->frame_size;
} while (s->pkt->size < s->period_size * s->frame_size);
av_packet_move_ref(pkt, s->pkt);
dts = av_gettime();
snd_pcm_delay(s->h, &delay);
dts -= av_rescale(delay + res, 1000000, s->sample_rate);
pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
s->last_period = res;
return 0;
}
static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
{
return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_CAPTURE);
}
static const AVOption options[] = {
{ "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ NULL },
};
static const AVClass alsa_demuxer_class = {
.class_name = "ALSA indev",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
};
const AVInputFormat ff_alsa_demuxer = {
.name = "alsa",
.long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
.priv_data_size = sizeof(AlsaData),
.read_header = audio_read_header,
.read_packet = audio_read_packet,
.read_close = ff_alsa_close,
.get_device_list = audio_get_device_list,
.flags = AVFMT_NOFILE,
.priv_class = &alsa_demuxer_class,
};
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